Download Part I: Introduction

CS 218 F 2003
Nov 3 lecture:



Streaming video/audio
Adaptive encoding (eg, layered encoding)
TCP friendliness
References:




J. Padhye, V.Firoiu, D. Towsley, J. Kurose” Modeling TCP
Throughput: a Simple Model and its Empirical Validation” Sigcomm
98
S. Floyd, J. Padhye, J.Widmer “Equation Based Congestion Control
for Unicast Applications”, Sigcomm 2000
Rejaie et al “ RAP: end to end control for real time streams”
Infocom 99
Tang et al “ RCS: Rate Control Scheme..for high bit error rates”
Streaming
 Important and growing application due to
reduction of storage costs, increase in high speed
net access from homes, enhancements to caching
and introduction of QoS in IP networks
 Audio/Video file is segmented and sent over
either TCP or UDP, public segmentation protocol:
Real-Time Protocol (RTP)
Using a Streaming Server
 This gets us around HTTP, allows a choice of UDP
vs. TCP and the application layer protocol can be
better tailored to Streaming; many enhancements
options are possible (see next slide)
Options When Using a Streaming
Server
 Use UDP, and Server sends at a rate (Compression and
Transmission) appropriate for client; to reduce jitter, Player
buffers initially for 2-5 seconds, then starts display
 Use TCP, and sender sends at maximum possible rate under
TCP; retransmit when error is encountered; Player uses a
much large buffer to smooth delivery rate of TCP
Real-Time Protocol (RTP)
 Provides standard packet format for real-time




application
Typically runs over UDP
Specifies header fields below
Payload Type: 7 bits, providing 128 possible
different types of encoding; eg PCM, MPEG2
video, etc.
Sequence Number: 16 bits; used to detect packet
loss
Real-Time Protocol (RTP)
 Timestamp: 32 bytes; gives the sampling instant
of the first audio/video byte in the packet; used
to remove jitter introduced by the network
 Synchronization Source identifier (SSRC): 32
bits; an id for the source of a stream; assigned
randomly by the source
RTP Control Protocol (RTCP)
 Protocol specifies report of packets exchanged between
sources and destinations of multimedia information
 Three reports are defined: Receiver reception, Sender, and
Source description
 Reports contain statistics such as the number of packets
sent, number of packets
lost, inter-arrival jitter
 Used to modify sender
transmission rates and
for diagnostics purposes